In a service for implementing peer-to-peer communication over the Internet, such as IP phones, videophones, or instant messengers, a signaling technology for controlling the establishment, change, and disconnection (or termination) of a peer-to-peer session (for example, communication session) over the Internet is important, in addition to the technology for transmitting, in real time, media information on audio, video, text or the like by using an IP (the Internet protocol) packet.
For example, IP phones, which have become widespread remarkably in recent years, are implemented by combining a VoIP (Voice over IP) technology for transmitting an audio signal in an IP packet in real time and the signaling technology.
For signaling protocols available for IP phones, H.323 methods recommended by ITU-T (International Telecommunication Union-Telecommunication sector) in 1997 and SIP (Session Initiation Protocol) provided as the standard track in RFC 3261, which was standardized by IETF (Internet Engineering Task Force) and issued in 2002, have been implemented. Especially, SIP, in which a message is described in text, is designed on the model of HTTP (Hyper Text Transfer Protocol) for web services and SMTP (Simple Message Transfer Protocol) for electronic mails. As a result, SIP is simple, highly scalable and highly compatible with the Internet and is now becoming the standard of the signaling protocols used in IP phones.
SIP is a signaling protocol for controlling the establishment, change, and disconnection (or termination) of a communication session between a pair of user terminals in an application layer.
The establishment, change and disconnection of a communication session is performed by exchanging a method (or a request message) and a response (response message) between user terminals through a relay server called SIP server deployed over the Internet, in accordance with predetermined steps.
For example, the establishment of a communication session starts from the transmission of INVITE message. The user terminal in SIP (or UA: User Agent) is identified on the basis of the URI (Uniform Resource Identifier) form, such as sip:hanako@fujitsu.com, and sip:taro@fujitsu.com.
By exchanging SIP messages, information on a medium to be used (audio, video or text), information on an encoding method to be used for an audio medium, information on a protocol to be used for transporting the audio packet, a port number to be used, an audio packet transmission cycle and so on are notified.
In a general telephone service, the reaction to the occurrence of a network fault and quick recovery of the service are essential for achieving the customer satisfaction and the reliability for the service. Upon occurrence of a fault, in order to properly respond to expected customer inquiries, to suppress the expansion of the influence, and to recovery the service, it has been desired to quickly grasp the range of influence (for example, influenced users).
In a conventional public switched telephone network: PSTN (Public Switched telephone Network), one line unit provides one communication path, and when a communication error or a congestion is detected, the communication path itself is blocked. Therefore, by checking the block state of the communication path, the range of influence can be identified.
An IP phone is a new form of telephone service and is basically a service based on IP protocol. Therefore, a technology for identifying the fault range on the basis of the concept of the communication path as in PSTN cannot be applicable thereto.
As a result, in reality, the fault detection in the IP phone service has no other choice than to rely on the fault detection method for IP networks. For example, the set of paths where a fault has occurred is identified by measuring the communication quality information on the combinations of all paths constructing the Internet between terminals, in the conventional technology as disclosed in Japanese Laid-open Patent Publication No. 2005-102180.
In the case of an IP network having a tree structure, connection information between network apparatus and user information accommodated in a network apparatus can be managed in association with each other in advance. In the case, when a fault has occurred an area of the network, all users accommodated in an area of the network which is closer to the user side than the area where the fault has occurred may be regarded as influenced by the fault.
In the case of an IP network having a mesh structure, a possible method is to narrow the range of influence by performing a communication test between contraposition area-pair in the network, for example, as disclosed in Japanese Laid-open Patent Publication No. 2005-102180.